[asterisk-users] SIp Signalling
Is there a way to force asterisk to take care only of sip signaling without forcing it to take care of rtp traffic? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime SIP
Probably I did not read well the information I am concerning, if I am going to use ARA for the SIP and I have register = user:secret:[EMAIL PROTECTED]:port/extension how I should input that line? If I am going to delete it from the DB I am forced to reload everything or there is a way to tell asterisk to remove only a particular entry? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.15 Voicemail
Hi after having installed asterisk 1.4.15 my voicemail does not work anymore. Am I the only one? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail hangup
Probably I was not able to explain myself properly however, for some measge this what happen -- Local/[EMAIL PROTECTED],2 Playing '/var/spool/asterisk/voicemail/default/300/Old/msg0003' (language 'it') == Spawn extension (servizi, , 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' I cannot listen the message and the voicemailmain exists I am using asterisk 1.4.13 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Fwd: Re: VoiceMail hangup]
Hi I have the same problem On Nov 13, 2007 9:10 AM, marcotasto [EMAIL PROTECTED] wrote: Hi Neofita, Doug and All. I think I've the same problem but I don't know if it's related to the bug suggested below. I try to explain my behavior: - I dial the voicemail extension. - I hear: You have 1 new message. Press 1 for new messages, press 2 for... or # to exit (I listen the complete message or most part of it) - I press 1 - I can hear the first recorded message. But, if: - I dial the voicemail extension. - I hear you have 1 new message. Press 1... 1 pressed (without waiting for the message playing) - Asterisk hangups. I'm not always able to replicate the problem but, as Il Neofita, I'm using the italian prompts... could be a problem related to that? Bye and regards Marco Signorini. Il Neofita wrote: -- Local/[EMAIL PROTECTED],2 Playing '/var/spool/asterisk/voicemail/default/300/Old/msg0003' (language 'it') == Spawn extension (servizi, , 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' It may be related to this bug: http://bugs.digium.com/view.php?id=11083 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoiceMail hangup
Hi, with some messages the voicemailmain after give me the information about the call (Days, hours and minutes) it finish. Whant can I check for solve this problem? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail hangup
Hi additional information if I am going to wait at least 3 seconds after the voicemail starts to give me the instruction I am able to listen my messages. But why I need to wait? On Nov 12, 2007 2:28 PM, Il Neofita [EMAIL PROTECTED] wrote: Hi, with some messages the voicemailmain after give me the information about the call (Days, hours and minutes) it finish. Whant can I check for solve this problem? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail hangup
Thank you for your answer. The problem is quite different for example, I am leaving a message of 5 seconds when I call to listen the message , asterisk answer and pass the call to voicemailmain and it plays the welcome message now if I press 1 before 3 or 4 seconds the voicemailmain gives me then information of the message send the command to play the message and it exists. If I wait more the 3 or 4 seconds and then I press 1 everything is going well for the same kind of message On Nov 12, 2007 3:53 PM, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Montag, den 12.11.2007, 15:14 -0500 schrieb Il Neofita: Hi additional information if I am going to wait at least 3 seconds after the voicemail starts to give me the instruction I am able to listen my messages. But why I need to wait? On Nov 12, 2007 2:28 PM, Il Neofita [EMAIL PROTECTED] wrote: Hi, with some messages the voicemailmain after give me the information about the call (Days, hours and minutes) it finish. Whant can I check for solve this problem? Read voicemail.conf. Look for minmessage setting - it will remove messages that are shorter than the given number of seconds. See http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail See http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf BR Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with loop counting?
Hi
I believe that
exten = s,7,GotoIf($[${trips}=4]?,8)
the , should be :
On 10/24/07, Phil Knighton [EMAIL PROTECTED] wrote:
Hi
I have a situation where I want to be able to count how many times a
caller goes round a loop of Please hold..., please continue to hold. I
have found an example on voip-info but I can't get it to work. Not sure if
I've got some syntax wrong somewhere? All that happens at the moment, is I
hit is the playback of som-debug at . Any ideas would be appreciated!
extensions.conf:
[so-mainmenu]
exten = s,1,Answer
exten = s,2,Set(trips=1)
exten = s,3,SetMusicOnHold(default)
exten = s,4,Set(TIMEOUT(digit)=5)
exten = s,5,Set(TIMEOUT(response)=10)
exten = s,6,Background(softopt/som-mainmenu)
exten = s,7,GotoIf($[${trips}=4]?,8)
exten = s,8,WaitExten(5)
exten = s,9,Wait(5)
exten = 1,1,Goto(so-sandm,s,1)
exten = 2,1,Goto(so-support,s,1)
exten = 3,1,Goto(so-accbill,s,1)
exten = 4,1,Goto(so-switchboard,s,1)
exten = 5,1,Goto(so-silentdial),s,1)
exten = s,10,Background(softopt/som-mainmenuretry)
exten = s,11,Wait(1)
exten = s,12,Background(softopt/som-mainmenuopts)
exten = s,13,Goto(s,7)
exten = ,1,Playback(softopt/som-debug)
exten = ,2,Hangup()
exten = i,1,Set(trips=$[${trips} + 1])
exten = i,2,Goto(s,7)
Cheers
Phil
Phil Knighton
Soft Option Technologies
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[asterisk-users] Force codec order
There is a way to force the order of the codecs in the sip.conf since the allow seams to let know only the accepted codec. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange behaviour afetr update from 1.2 to 1.4
Hi, I update from asterisk 1.2 to 1.4 and I have some problems. In the extensions I used DIAL(SIP/100SIP/101,30,tTr) if I receive a call from an external providers now in 1.4 I recieve only one ring What can I do to solve this problem? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GTALK problem
Hi, I installed gtalk on asterisk 1.4.12.1, I change on rtp.conf the port from 1000 to 4 If I calling asterisk with GTALK in english everything is ok, however, some of my friends with the italian version of gtalk they cannot have the audio. Is it a bug? Or I did some mistake ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GTALK problem
Thank you I need to wait the international version of gtalk On 10/11/07, Philippe Sultan [EMAIL PROTECTED] wrote: If I calling asterisk with GTALK in english everything is ok, however, some of my friends with the italian version of gtalk they cannot have the audio. Audio problems might be experienced with older Gtalk clients. Version 1.0.0.104 is reported to work. The following resources may help you : http://www.voip-info.org/wiki/view/Asterisk+Google+Talk#Bugsampknownissues http://bugs.digium.com/view.php?id=10512 Hope this will help you solve the problem, Philippe ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Curiosity Max Calls
Hi is there a tool to know what was the maximum calls that asterisk managed? Thank you ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call hangup after 60seconds
Hi, I have a client (xlite) connected to my server, on the server I have type=friend and siptimeout=60, canreinvite=yes and dial with tT option, the server is listening on port 5060. However, xlite is connect to a router where the port 5060 is blocked, therefore, I am using 5065 and I have an iptables rule to transfer the incoming packet from 5065 to 5060, I cannot use the port 5065 since some ATA the do not allow the change of the port. When I am calling with xlite the call endup after 60seconds, but in the 60seconds I can talk. Now if I am setting the client (in the sip.conf) in peer everything is working. Someone can explain to me why? What I am doing wrong? Thank you ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Behaviour
Thank you I will try tonight On 9/10/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Montag, den 10.09.2007, 05:14 +0200 schrieb Il Neofita: On 9/9/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita: Well, it seems there are differences between those accounts then. You might want to post your sip.conf, and -if that is possible- the ATA conf file; or at least a writedown of the configuration there. First of all, thank you for you reply The ATA is the Fritz!Box and I tried with different FW version but I have the same behaviour I have been using FritzBoxes for quite a while, and have not found such strange bugs - except after a Firmware Upgrade. It seems after some upgrades you need to do a factory reset (via the web interface) and enter your data again, else they behave stupidly. this is part of the sip.conf [180] type=peer username=180 secret=aa callerid=First180 canreinvite = yes host = dynamic dtmfmode = rfc2833 qualify = yes nat = yes context = mycont disallow = all allow = g726 allow = g723 allow = ulaw allow = alaw allow = g729 allow = gsm [181] type=peer username=181 secret=bb callerid=Second181 canreinvite = yes host = dynamic dtmfmode = rfc2833 qualify = yes nat = yes context = mycont disallow = all allow = g726 allow = g723 allow = ulaw allow = alaw allow = g729 allow = gsm Looks pretty OK to me. Just a stupid idea: Do you have a [general] section before those two? And then, I use type=friend, not type=peer, that _might_ make a difference in how asterisk matches sip.conf contexts to registered clients. 8 From my sip.conf: [sip501] mailbox=01 callerid=501 type=friend username=sip501 secret=lk1j2eu89 context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm allow=ulaw [sip502] mailbox=02 callerid=502 type=friend username=sip502 secret=1092jd0 context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm allow=ulaw =8 Note: Those two accounts belong to the same FritzBox. I tried to switch the account for the two ports but what it is important is only the order in the sip.conf That made me think about that friend/peer thingy. I found some information in german and I do not know it The FritzBoxes are popular here in Germany - no wonder, being a German manufactured product and being given away for (nearly) free with any 2-year DSL contract... I like them nevertheless :) BR, HTH Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange Behaviour
Hi, my ATA has two phones attached and the possibility to set different accounts. I put two account of my asterisk server, however, it is able to call only with the second one in order to the sip.conf and the first it gives me 403. And idea how to solve it? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Behaviour
On 9/9/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita: Well, it seems there are differences between those accounts then. You might want to post your sip.conf, and -if that is possible- the ATA conf file; or at least a writedown of the configuration there. First of all, thank you for you reply The ATA is the Fritz!Box and I tried with different FW version but I have the same behaviour this is part of the sip.conf [180] type=peer username=180 secret=aa callerid=First180 canreinvite = yes host = dynamic dtmfmode = rfc2833 qualify = yes nat = yes context = mycont disallow = all allow = g726 allow = g723 allow = ulaw allow = alaw allow = g729 allow = gsm [181] type=peer username=181 secret=bb callerid=Second181 canreinvite = yes host = dynamic dtmfmode = rfc2833 qualify = yes nat = yes context = mycont disallow = all allow = g726 allow = g723 allow = ulaw allow = alaw allow = g729 allow = gsm If those are not the source of trouble, _I_ probably would switch the accounts in the ATA (port A versus port B) and try if the problem sticks with the port or with the account. I tried to switch the account for the two ports but what it is important is only the order in the sip.conf I would also google if there are known problems with my ATA, look if a newer firmware is available, if there are informative messages that are worth a verbatim quote, and get another bottle of beer to keep the sunday relaxation at a proper level. I found some information in german and I do not know it ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extra field
Hi, I am using my asterisk server like a gateway and one provider ask me to pass an extra field with the IP of the peer that is using the connection, probably to have more control on the authentication. I was wondering how I can implement this. Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400p Loaded only once
Hi
when I turn on my PC I able to load the drivers and start my card,
if I reboot the PC I have the following error
ztcfg -vvv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
4 channels configured.
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
This is part of my dmesg
audit(1172747900.510:5): avc: denied { net_bind_service } for pid=1657
comm=hidd capability=10 scontext=system_u:system_r:bluetooth_t:s0
tcontext=system_u:system_r:bluetooth_t:s0 tclass=capability
SELinux: initialized (dev autofs, type autofs), uses genfs_contexts
eth0: no IPv6 routers present
[drm] Initialized drm 1.1.0 20060810
ACPI: PCI Interrupt :01:00.0[A] - GSI 16 (level, low) - IRQ 19
[drm] Initialized r128 2.5.0 20030725 on minor 0
agpgart: Found an AGP 3.0 compliant device at :00:00.0.
agpgart: Device is in legacy mode, falling back to 2.x
agpgart: Putting AGP V2 device at :00:00.0 into 1x mode
agpgart: Putting AGP V2 device at :01:00.0 into 1x mode
audit(1172747921.184:6): avc: denied { getattr } for pid=2323
comm=pam_console_app name=card0 dev=tmpfs ino=7969
scontext=system_u:system_r:pam_console_t:s0-s0:c0.c255tcontext=system_u:object_r:device_t:s0
tclass=chr_file
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.14
Zaptel Echo Canceller: KB1
and finally this is my configuration
fxsks=1-4
loadzone=us
defaultzone=us
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Re: [asterisk-users] TDM400p Loaded only once
Thank you for the answer
after
modprobe wctdm
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
FATAL: Error running install command for wctdm
/proc/zaptel/ (empty)
/usr/src/asterisk/zaptel-1.2.14/xpp/utils/genzaptelconf -l (no result)
On 3/1/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Mar 01, 2007 at 06:21:38AM -0500, Il Neofita wrote:
Hi
when I turn on my PC I able to load the drivers and start my card,
if I reboot the PC I have the following error
ztcfg -vvv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
This is what you attempt to configure
4 channels configured.
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
And that is the result.
What do you see on /proc/zaptel ?
What is the output of 'xpp/utils/genzaptelconf -l' from the zaptel
source directory?
This is part of my dmesg
audit(1172747900.510:5): avc: denied { net_bind_service }
for pid=1657
comm=hidd capability=10 scontext=system_u:system_r:bluetooth_t:s0
tcontext=system_u:system_r:bluetooth_t:s0 tclass=capability
SELinux: initialized (dev autofs, type autofs), uses genfs_contexts
eth0: no IPv6 routers present
[drm] Initialized drm 1.1.0 20060810
ACPI: PCI Interrupt :01:00.0[A] - GSI 16 (level, low) - IRQ 19
[drm] Initialized r128 2.5.0 20030725 on minor 0
agpgart: Found an AGP 3.0 compliant device at :00:00.0.
agpgart: Device is in legacy mode, falling back to 2.x
agpgart: Putting AGP V2 device at :00:00.0 into 1x mode
agpgart: Putting AGP V2 device at :01:00.0 into 1x mode
audit(1172747921.184:6): avc: denied { getattr } for pid=2323
comm=pam_console_app name=card0 dev=tmpfs ino=7969
scontext=system_u:system_r:pam_console_t:s0-s0:
c0.c255tcontext=system_u:object_r:device_t:s0
tclass=chr_file
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.14
Zaptel Echo Canceller: KB1
Load of zaptel. No load of wctdm in sight.
and finally this is my configuration
fxsks=1-4
loadzone=us
defaultzone=us
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icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Strange Noise
Hi, today with my asterisk during a call I had a very strange noise, it was the typical noise that you can have when your device uses a bad power supply. I change phone and I had the same behavior after I while I tried again and the noise was disappeared. What can I check? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange behaviour with Dial cmd
Yes, but I would like to try a number and after to try a second one. Any Idea how to avoid this. On 2/18/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: C F wrote: Asterisk supports this directly by issuing the hangup command before the answer command. However, when using an analog interface like FXO the line has no way of knowing you just hung up and will continue to ring, which asterisk will see as a new call. in my experience even when using a PRI if i dont give the pri cause the provider re initiates the call. This would only happen if you blindly run two Dial lines in sequence in your dialplan. Don't do this. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange behaviour with Dial cmd
On 2/14/07, Yuan LIU [EMAIL PROTECTED] wrote: Well, you'll have to decide how you want to hang up the caller: Do you want him/her to be ignored, or to be told that you are not available (like an answering machine)? You also need to tell Asterisk how to determine if the next invite comes from the same caller during the same session. (These are not very easy tasks but doable.) In either case, you need to add a flag to your dial plan, set it after it rings your cell, and reset it once Asterisk determines that THIS caller has been hung up. (Of course you can do what Vacation does in E-mail: set up a flag for each identifiable caller, and only call your mobile once until you reset them all. The algorithm would be simpler but more unidentifiable callers will be ignored.) Your dial plan will check this flag before ringing your cell, then branch accordingly. Hope this helps. Yuan Liu Thank you for the answer. Right know I solved sending after everything to the voicemail. But I tought that hangup was suppose to close the call, however, is not the case and a really did not catch why. I will try to read a bit more. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange behaviour with Dial cmd
Ok thank you a lot!!! On 2/15/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Il Neofita [EMAIL PROTECTED] Date: Thu, 15 Feb 2007 03:37:14 -0500 But I tought that hangup was suppose to close the call, however, is not the case and a really did not catch why. Now I see where the confusion comes from. Asterisk doesn't really speak English - or Chinese for that matter:-) In telephony, there is no way for the callee to tell the caller to stop ringing - unless you answer it first. Once you answer, you can do a number of things, the rudest being to immediately hang up. (I saw live people doing this intentionally.) Your only other option really is to ignore. I just thought up this simple method to ignore: divert the dial plan to simply Wait() an unreasonable amount of time in hope that the caller hangs up. exten = s,1,Dial(yourcell,5) exten = s,n,Wait(300) That's assuming your provider provides disconnect supervision. You can also Play(prerecorded,noanswer) if your provider supports it. (Won't hurt to try even if not - but disconnect supervision is a must.) Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange behaviour with Dial cmd
I have this simple context I am register to an external provider and when I am not home I would like to transfer the phone outside The problem that the call goes in loop I cannot understand why. Can you figure out my error? Thank you sip.conf register = user:[EMAIL PROTECTED]/400 [inside] exten = _4X.,1,dial(SIP/ext_400_124/555123,5,tT) exten = _4X.,2,hangup -- Executing Dial(SIP/555123-081b8eb0, SIP/ext_400_124/555123|5|tT) in new stack -- Called ext_400_124/555123 -- SIP/ext_400_124-081bf848 is ringing -- Nobody picked up in 5000 ms -- Executing Hangup(SIP/555123-081b8eb0, ) in new stack == Spawn extension (inside, 5123, 2) exited non-zero on 'SIP/555123-081b8eb0' -- Executing Dial(SIP/555123-081c79a8, SIP/ext_400_124/555123|5|tT) in new stack -- Called ext_400_124/555123 -- SIP/ext_400_124-081ccee8 is ringing == Spawn extension (inside, 5123, 1) exited non-zero on 'SIP/555123-081c79a8' ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange behaviour with Dial cmd
This is the situation A call me at my provider 1 I am not home and I would like to transfer the call I do not pickup the call for some reason I would like to hangup the caller, however, my asterisk try again to call on my mobile over and over I would like to stop it. Any idea? Thank you a lot. On 2/14/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Il Neofita [EMAIL PROTECTED] Date: Wed, 14 Feb 2007 19:30:51 -0500 I have this simple context I am register to an external provider and when I am not home I would like to transfer the phone outside The problem that the call goes in loop I don't see any loop in records below? What your dial plan does is what you told it to: ring for 5 sec, then hang up. Because the calling party hasn't given up, your dial plan rings again. Where's the error? Yuan Liu I cannot understand why. Can you figure out my error? Thank you sip.conf register = user:[EMAIL PROTECTED]/400 [inside] exten = _4X.,1,dial(SIP/ext_400_124/555123,5,tT) exten = _4X.,2,hangup -- Executing Dial(SIP/555123-081b8eb0, SIP/ext_400_124/555123|5|tT) in new stack -- Called ext_400_124/555123 -- SIP/ext_400_124-081bf848 is ringing -- Nobody picked up in 5000 ms -- Executing Hangup(SIP/555123-081b8eb0, ) in new stack == Spawn extension (inside, 5123, 2) exited non-zero on 'SIP/555123-081b8eb0' -- Executing Dial(SIP/555123-081c79a8, SIP/ext_400_124/555123|5|tT) in new stack -- Called ext_400_124/555123 -- SIP/ext_400_124-081ccee8 is ringing == Spawn extension (inside, 5123, 1) exited non-zero on 'SIP/555123-081c79a8' ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4 FC5 and Gtalk
Sei riuscito? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: asterisk 1.4 FC5 and Gtalk
Hi, I tried thousands of time and finally I am a step closer to the solution. I recompile iksemel with the option --prefix=/usr I erase my zaptel-1.4, asterisk-1.4 and asterisk-addons-1.4, re-extracting everything from the tar recompile everything and now jabber is working or almost. When I received a call asterisk crash, but I saw that there is a patch and I am applying that right know. See you On 2/9/07, Mani Sridhar [EMAIL PROTECTED] wrote: i saw the same problem and here is a thread where i mentioned how i fixed it.. http://lists.digium.com/pipermail/asterisk-users/2006-November/171783.html look for my previous mails in this thread sometime september-november 2006 . btw, i can't get asterisk to work with google talk yet. thanks sridhar From: [EMAIL PROTECTED] Reply-To: asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 31, Issue 39 Date: Fri, 9 Feb 2007 18:30:38 -0700 (MST) MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc11-f20.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Fri, 9 Feb 2007 17:36:29 -0800 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 1C63E2FCC32;Fri, 9 Feb 2007 18:30:38 -0700 (MST) X-Message-Info: txF49lGdW41RYrP+Tdoh49JHbOTLdMhagyFw1S6VRR0= X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto: [EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto: [EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 10 Feb 2007 01:36:30.0423 (UTC) FILETIME=[E3926E70:01C74CB3] Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Message: 12 Date: Fri, 9 Feb 2007 23:28:15 +0100 From: marcotasto [EMAIL PROTECTED] Subject: Re: [asterisk-users] asterisk 1.4 FC5 and Gtalk To: asterisk-users asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Ciao Neofita. I'm trying my GTalk account and I'm still having the same problem. I've installed the gnuTLS-developer rpms and rebuilt and re-installed the complete Asterisk package but without success. I'm working with OpenSuse 10.2. This is my debug info that's quite similar to what you've posted: JABBER: asterisk OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='gmail.com' version='1.0' Gateway*CLI JABBER: asterisk INCOMING: ?xml version=1.0 encoding=UTF-8?stream:stream from=gmail.com id=2601C222D846D6C3 version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:clientstream:featuresstarttls xmlns=urn:ietf:params:xml:ns:xmpp-tls/mechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismX-GOOGLE-TOKEN/mechanism/mechanisms/stream:features [Feb 9 23:15:43] ERROR[24214]: res_jabber.c:482 aji_act_hook: gnuTLS not installed. There is someone knowing what's the problem and that could help us? Best regards, Marco Signorini. -- Passa a Infostrada. ADSL e Telefono senza limiti e senza canone Telecom http://click.libero.it/infostrada9feb07 _ Gossips, movie reviews, photogallery and more http://content.msn.co.in/Entertainment/Default ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: asterisk 1.4 FC5 and Gtalk
On 2/10/07, Patrick [EMAIL PROTECTED] wrote: Where can I find that patch? Thanks, Patrick Hi Patrick, I downloaded the patch from here http://bugs.digium.com/view.php?id=7764 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chan_Cellphone
Hi, I download the last svn and I also look around but I cannot find the source, I only found the patch http://bugs.digium.com/print_bug_page.php?bug_id=8919 any one can help me out. thx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BindPort
The point is to use more than one port, I think the only way is to use the redirect from iptables On 2/6/07, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Ciao, just change port value in sip.conf. Giorgio Il Neofita wrote: Hi, I was wondering if it is possible to set asterisk in order to listen to different ports for the sip or I need to do this operation with iptables? All of this since some time the port 5060 is blocked. Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Chan_Cellphone
I start the patch and automatically created the file. But now on the menu I cannot select chan_cellphone I launched ./bootstrap.sh and after ./configure in my /usr/include/bluetooth I have the header but I cannot select the option any idea? On 2/9/07, Il Neofita [EMAIL PROTECTED] wrote: Hi, I download the last svn and I also look around but I cannot find the source, I only found the patch http://bugs.digium.com/print_bug_page.php?bug_id=8919 any one can help me out. thx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4 FC5 and Gtalk
JABBER: gtalk_account OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to=' gmail.com' version='1.0' localhost*CLI jabber show tes JABBER: gtalk_account INCOMING: ?xml version=1.0 encoding=UTF-8?stream:stream from=gmail.com id=58D5EEFB06C20E13 version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:client [Feb 9 21:11:15] ERROR[2061]: res_jabber.c:482 aji_act_hook: gnuTLS not installed. I installed all the gnutls but I still have this error [EMAIL PROTECTED] ~]# rpm -qa | grep gnutls gnutls-utils-1.2.10-3 gnutls-1.2.10-3 gnutls-devel-1.2.10-3 Do you know how to solve it? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BindPort
Hi, I was wondering if it is possible to set asterisk in order to listen to different ports for the sip or I need to do this operation with iptables? All of this since some time the port 5060 is blocked. Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chanskype
Hi, I was wondering if someone had problems with chanskype. Since I am wondering if they are a credible company or not. See you On 1/20/07, Moises Silva [EMAIL PROTECTED] wrote: Hi, I tried the try version of chanskype, however, everytime that I make a call asterisk generate an error So you think is easy to us guess wich error you are getting? Seriously, I think you should read this: http://www.catb.org/~esr/faqs/smart-questions.html Anyone has experience with this? Since I tried to contact the support but they never replied. Please provide more information about the error, and search in voip-info.org how to raise the verbosity level of asterisk in the console. Also sometimes help searching the error in google. Regards -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chanskype
Hi, I tried the try version of chanskype, however, everytime that I make a call asterisk generate an error Anyone has experience with this? Since I tried to contact the support but they never replied. Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange error
Someone know why my asterisk gives me the following msgs? Thank you - Got SIP response 603 Declined (no dialog) back from X.X.X.Xhttp://82.51.224.34/ -- Got SIP response 603 Declined (no dialog) back from X.X.X.Yhttp://82.51.224.34/http://82.104.4.192/http://82.104.4.192/ -- Got SIP response 603 Declined (no dialog) back from X.X.X.Zhttp://82.51.224.34/http://82.51.224.34/ -- Got SIP response 603 Declined (no dialog) back from X.X.X.Xhttp://82.51.224.34/http://82.104.4.192/ -- Got SIP response 603 Declined (no dialog) back from X.X.X.Xhttp://82.51.224.34/http://82.51.224.34/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for toll free in Italy
I do not think that there are some company that offer a toll free number (Numero verde in italian) But contact on of these three providers http://www.eutelia.it/tlc/ http://www.unidata.it/ http://messagenet.it/ If they have one of these should be able to give to you See you On 1/8/07, CM Rahman [EMAIL PROTECTED] wrote: *Hi,* ** *I am looking for tollfree number in italy. Anybody providing that? Charge per minute? It will connect to my asterisk pbx box.* ** *Thanks* ** *CM* __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WebPhone
On 7/3/06, Jean-Denis Girard [EMAIL PROTECTED] wrote: But I'm not sure that MozPhone is what the original poster asked.No, however, I like to read all these different point of view. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WebPhone
Hi,someone know a good webphone, possibily a free oneThx ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Canreinvite
I will try your suggestion and I will let you know. Thank you On 6/18/06, Philippe Lindheimer [EMAIL PROTECTED] wrote:How have you confirmed that they did not reinvite? The channels are still controlled by Asterisk (sip signalling), it is the rtp streams that go direct. You can do a sip show channel 146b518a4cd on the specific channel to see where the rtp streams are going. Or ... if this is the only active channel on the box, just do a rtp debug. If the rtp stream is going through asterisk, it will be very obvious. If not, you won't see a constant flow of rtp debug messages going on. pFrom: Il Neofita [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate: Sun, 18 Jun 2006 05:01:20 -0400Subject: Re: [Asterisk-Users] Canreinvite This is the dial in extensionsexten = _40001,1,Dial(SIP/40001,30) exten = _40002,1,Dial(SIP/40002,30) From: Il Neofita [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sun, 18 Jun 2006 05:22:35 -0400Subject: Re: [Asterisk-Users] Canreinvite cosa vedo a console -- Executing Dial(SIP/40001-3760, SIP/40002|30) in new stack -- Called 40002 -- SIP/40002-4753 is ringing -- SIP/40002-4753 answered SIP/40001-3760 -- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753srvlinux*CLI sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message82.X2.XX3.X3 40002 146b518a4cd 00103/0 alaw No Tx: ACK 82.X2.XX3.X3 40001 CBD1DB85-8B 00102/30987 alaw No Tx: ACK2 active SIP channels Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Canreinvite
This is the dial in extensionsexten = _40001,1,Dial(SIP/40001,30) exten = _40002,1,Dial(SIP/40002,30) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Canreinvite
cosa vedo a console -- Executing Dial(SIP/40001-3760, SIP/40002|30) in new stack -- Called 40002 -- SIP/40002-4753 is ringing -- SIP/40002-4753 answered SIP/40001-3760 -- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753srvlinux*CLI sip show channelsPeer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message82.X2.XX3.X3 40002 146b518a4cd 00103/0 alaw No Tx: ACK 82.X2.XX3.X3 40001 CBD1DB85-8B 00102/30987 alaw No Tx: ACK2 active SIP channels ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Canreinvite
I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel Module.symvers missing
Hi,when I am going to compile the zaptel I receive this messageake -C /lib/modules/2.6.12-12mdk/build SUBDIRS=/usr/src/asterisk/zaptel-1.2.5 XPPMOD= modulesmake[1]: Entering directory `/usr/src/linux-2.6.12-12mdk ' WARNING: Symbol version dump /usr/src/linux-2.6.12-12mdk/Module.symvers is missing; modules will have no dependencies and modversions...lrwxrwxrwx 1 root root 18 mag 23 17:42 linux - linux-2.6.12-12mdk/drwxr-xr-x 22 root root 4096 mag 23 18:04 linux-2.6.12-12mdk/..Any idea how to generate this file?cat /proc/versionLinux version 2.6.12-12mdk ( [EMAIL PROTECTED]) (gcc version 4.0.1 (4.0.1-5mdk for Mandriva Linux release 2006.0)) #1 Fri Sep 9 18:15:22 CEST 2005 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NOTIFY Problem
Hi,one of my WiFI phone has problem with the notify asterisk signal to me the following Apr 29 06:49:16 WARNING[6455] chan_sip.c: Host '192.168.100.124' does not implement 'NOTIFY' In theory the phone support this function.Any idea? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NOTIFY Problem
I agree with you, but I would like to find a way to use the notification. I tough that there was a work around.On 4/29/06, tom [EMAIL PROTECTED] wrote:Il Neofita wrote: Hi, one of my WiFI phone has problem with the notify asterisk signal to me the following Apr 29 06:49:16 WARNING[6455] chan_sip.c: Host '*MailScanner warning: numerical links are often malicious:* 192.168.100.124 http://192.168.100.124' does not implement 'NOTIFY' In theory the phone support this function. Any idea?If you remove the mailbox= bit of sip.conf for that host, then asterisk will almost never send it a notify.Either way, is it causing a problem?___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to use the cmd SMS
Hi,I try to use my phone that has a SMS capability with asterisk.I am not able to receive SMS, someone can help me out?This what I am able to have but nothing more-- Executing SMS(SIP/503-7d2e, 508|sa) in new stack -- SMS TX 93 00 6D ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Remote UNIX connection disconnected over and over
Hi,I am pretty sure that you already answer to this question, but I was not able to find the solutionon the console I have over and over the following msgs-- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected Any idea? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_gsm_bt Impression
Hi,Has anyone proved the chan_gsm_bt ??Any impression? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error Header field Via
Someone know the meaing of this error?chan_sip.c:3853 copy_via_headers: No header field 'Via' present to copy ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 on way voice
This is my debugtried with netmeeting I can still hear but when I talk nothing happenmygw*CLI h.323 debugH323 debug enabledmygw== New H.323 Connection created. -- Setting up Call -- oCall token: [ip$192.168.1.219:1057/8226] -- oCalling party name: [myPersonal] -- oCalling party number: [] -- oCalled party name: [ip$192.168.1.214:1720] -- oCalled party number: []mygw--Received SETUP messageAllowed Codecs: mygw Table: G.729A 1 G.729 2 G.723.1 3 G.711-uLaw-64k 4 G.711-ALaw-64k 5 UserInput/hookflash 6 UserInput/RFC2833 7 Set:aco*CLI 0:aco*CLI 0:o*CLI G.729A 1 G.729 2 G.723.1 3 G.711-uLaw-64k 4 G.711-ALaw-64k 5 1:o*CLI UserInput/hookflash 6 2:o*CLI UserInput/RFC2833 7mygw*CLImygw=-= In OnAnswerCall for call 8226mygw*CLI - Progress Indicator: 0mygw*CLI - Inserting PI of 0 into ALERTING message == Starting H323/ip$192.168.1.219:1057/8226 at default,ip$192.168.1.214:1720,1 failed so falling back to exten 's' -- Executing Playback(H323/ip$192.168.1.219:1057/8226, demo-echotest) in new stack mygwAnswering call ip$192.168.1.219:1057/8226 -- Playing 'demo-echotest' (language 'en')mygw-- Started logical channel: sending G.711-uLaw-64kmygw*CLI -- channelsOpen = 1mygw=-= In OnConnectionEstablished for call 8226 mygw*CLI -- Connection Established with myPersonal [192.168.1.219]mygwMyH323_ExternalRTPChannel::OnReceivedAckPDUmygw*CLI -- remoteIpAddress: 192.168.1.219 -- remotePort: 49600 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 192.168.1.219 -- remotePort: 49600 -- ExternalIpAddress: 192.168.1.214 -- ExternalPort: 17950mygw*CLI . -- Executing Echo(H323/ip$192.168.1.219:1057/8226, ) in new stack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codec Problem
I have the license for G729, however I need to use a different codec for the prepaid service, but when the call is started I have this errorAsked to transmit frame type 256, while native formats is 4 (read/write = 4/4) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8446b50', 10 retries!
I never so this error.I am using H323 with Asterisk 1.2.6 Any idea what can be the problem? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 on way voice
Hi,I installed H323, however when I make a call from SIP Phone - Asterisk H323 - Provider H323 the provider can hear me, but I cannot hear nothing.The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP. Any thoughts? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 behind a Firewall
There is a proble to put an H323 Asterisk server behind an iptables firewall? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 Info
Hi,I compiled for my asterisk 1.2.4 the openh323but when I give this commandh.323 show codecsI receive thisAllowed Codecs: Table:Set:I cannot test with msn if everything is working since I am outside and I cannot access to the firewall. Someone can tell me if I need to install the oh323 since I do not know if I need it or notThank you ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AstCC
Hi,I am wondering if it is possible with astcc to make a second call without hangup and be oblige to re-enter all the codes.Any idea how to do?Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P busy
Hi,I have a TDM400P with 4 FXO The TDM after it receive a call do not hangup properly, it takes the line occupied. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P busy
Yes, I changed.Thank youOn 3/27/06, Infobox Peru [EMAIL PROTECTED] wrote: Maybe your lines use polarity reversals for hangup detection.On 3/27/06, Il Neofita [EMAIL PROTECTED] wrote: Hi,I have a TDM400P with 4 FXO The TDM after it receive a call do not hangup properly, it takes the line occupied. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MWI problem
Sorry if I am always here asking for MWI, but I do not know how to solve this issue, I have my ATAs (Azatel 200 and Fritz!Box) that they think that I have a message waiting. Anyone knows how to solve this issue? Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WMI Problem
Hi, I was not able to find any indication for this problem that I have right now. My phones connected to an Azatel 200 they always indicate that I have a message waiting to be listen. However, I do not have any message. I also checked using the console show voicemail user for context but I have 0 messages. Any idea what I need to check? I am using Asterisk 1.2.1 Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA does not register
I am not able to register an external ATA on my asterisk 2.0 Beta This is the debug Any idea? server01*CLI -- SIP read from CLIENTIP:5060: REGISTER sip:SIPSERVERIP SIP/2.0 Via: SIP/2.0/UDP PRIVATEIP;rport;branch=z9hG4bK455E4AEA9A9954FB135D6D788DA2 From: sip:[EMAIL PROTECTED];tag=1564789518 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Max-Forwards: 70 User-Agent: AVM FRITZ!Box Fon ata 11.03.37 Supported: 100rel, replaces Allow-Events: telephone-event Allow-Events: refer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, PRACK, INFO, SUBSCRIBE, NOTIFY, REFER Accept: application/sdp Accept-Encoding: identity Content-Length: 0 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WMI problem
I installed astersik 1.2beta and from that point the led that indicate a new call flash. The ATA installed is an AZATEL. Any idea what can I check? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TXFAX
Hi, do you know what it means the following: Call failed to go through, reason 3 I received it when I try to send a FAX and no one answer it. Thank you ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sending fax
HI Chris I am interested, I would like to know how I can have the opportunity to test your program. On 9/9/05, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Sep 08, 2005 at 06:42:49PM -0500, Chris Shipman wrote: I started working on a program using Ghostscript and Redmon to generate the tif in windows by a printer. So far I am using FTP to transfer the tiff and call file.At least until I figure something better out.Why don't you look at IPP (Internet Printing Protocol)? a protocol forsubmitting jobs over HTTP of some sort. Server is already implemented ine.g. cups.HTTP allows a nice header with some extra fields. I wonder if that can be abused to get the call information through. (and am I re-inventingsome wheels in the process?)--Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's[EMAIL PROTECTED] | |bestICQ# 16849755 | | friend___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Txfax
This is the script I do not remeber the name of the author For me is working On 9/9/05, Chris Shipman [EMAIL PROTECTED] wrote: Thanks Luki,Maybe it was the wrong place to mention it.Back to the other matter: The program will create a PDF or tif. Itcan submit the fax by FTP or by AstFax.If someone is willing to develop a CGI/PERL script for HTTP submissions I will integrate that into the program as well.Since my program uses a lot of Open source programs I will release itunder the GPL as well.I plan to test the faxing portionsnext week.After I have some of the bugs worked out I will release the program.Regards,Chris- Original Message -From: Luki [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSent: Friday, September 09, 2005 6:20 PMSubject: Re: [Asterisk-Users] Txfax I don't understand, what you are telling or asking us with this information. Has it something to do with your question? But others might understand... I do. Yes, it's related. Essentially it's the equivalent of WHFC for Hylafax -- a Windows client that will allow Windows users print from any program to a virtual fax printer; Chris's program will then submit the job as a TIFF to the asterisk server along with a call file. Here's the relevant part: it's quite important to know if the fax went through or not, and if it should be rescheduled for resending without (Windows) user intervention. Anyway, IMO Roger's hack is useful regardless and I'd be interested in trying Chris's solution out once it's ready (and if he chooses to share it). --Luki ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users asterfax.0.22.tar.gz Description: GNU Zip compressed data ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extension a
Hi, I would like to use the * when I am in the asnwer machine, but I received a message asking for the temporary pass code. Where I need to put this pass? I am using asterisk 1.2.0 beta 1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Txfax
Is theresome way to know if the fax was received correctly or not? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sending fax
The site is in italian, and you need to register your self in order to download the script. The script is in perl and you need to start in order to simulate the hylafax daemon after that you can use WHFC On 9/8/05, Matthew Gibson [EMAIL PROTECTED] wrote: Sorry to interrupt :)But I believe what you guys are searching for lays here:http://www.inter7.com/?page=astfaxThanks,MattWiley Siler wrote: Google can translate if that helps... w___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sending fax
Hi,
I found on a forum a script that emulate a hylafax this is the linkhttp://www.vocesuip.com/viewtopic.php?p=2423
You can use the WHFC in order to send a fax to asterisk.
On 9/5/05, Harald Klein [EMAIL PROTECTED] wrote:
Hi Chris, Hi Arne,Am 5.9.2005 schrieb Chris Shipman [EMAIL PROTECTED]:I'veseen some programs that install as a printer and create an image.
However this would be to cumbersome for your average user.It would need to be able to print to as local printer and then send outAsterisk.What about:Client with Postscript printer driver
Some kind of a printing system (samba with lpr[ng] and/or cups etc.) toaccess the fax-printer via smb/cifs/lpr/ipp/whatever..Output filter for the fax-printer to convert Postscript to tiff andgeneratea call file with App txfax...
The problem is to tell the printer the number to fax to...You can grep in the Postscript file for a predefined string (for exampleFax Recpient Nr) and generate some matching templates in your office
suite..Search for HylaFax solutions, they are pretty much the same...HariChris- Original Message -From: Arne Morten Johansen
[EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comSent: Monday, September 05, 2005 6:27 AM
Subject: SV: [Asterisk-Users] sending fax What about faxing yourself if you don't have a scanner? -Opprinnelig melding- Fra:
[EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] På vegne av Johan vanTongeren Sendt: 5. september 2005 09:11
Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [Asterisk-Users] sending fax [macro-fax-dialing] exten = s,1,SetCIDNum(0${CALLERIDNUM})
exten = s,2,Dial(Zap/g${ARG2}/${ARG1},20,,t) exten = s,3,Goto(900) exten = s,103,Goto(900) exten = s,900,Busy exten = s,901,Hangup
-Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] Namens Chris Shipman Verzonden: maandag 5 september 2005 7:22 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] sending fax
I've read alot on the wiki about sending and receiving faxes thru asterisk. I've gotten the receive to work great.My question is how does one send a fax?
I see lots of instructions about how to send the image to asterisk by email, etc.The problem is how doesone make the image of the fax to begin with? Has anyone come up with a good solution for this?
Regards, Chris ___ --Bandwidth and Colocation sponsored by
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Re: [Asterisk-Users] Asterisk architecture
>From my point of view I do not see any issue with that scenario. On 9/5/05, housi mueller [EMAIL PROTECTED] wrote: I am new with asterisk and hope somebody can help me. Is a configuration like shown on the picture with asterisk correct? Some phone calls arriving in Branch 1 should be redirected automatically to Branch 2 and all phone calls made from Branch 2 should going out over Branch 1. (Branch 2 is not connected directly with a PSTN.) Thank you in advace Housi Mueller Click here to donate to the Hurricane Katrina relief effort. ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk architecture
The call generate from branch2 can be send to the asterisk in Branch1 with a trunk the same think the call received from branch1 the only thing that is not cleat how you want transfer automatically the call received from the pstn. What rule you want use?On 9/5/05, housi mueller [EMAIL PROTECTED] wrote: Why not? How would you solve then the Brench1/Branch2 issue?? a [EMAIL PROTECTED] wrote: From my point of view I do not see any issue with that scenario. On 9/5/05, housi mueller [EMAIL PROTECTED] wrote: I am new with asterisk and hope somebody can help me. Is a configuration like shown on the picture with asterisk correct? Some phone calls arriving in Branch 1 should be redirected automatically to Branch 2 and all phone calls made from Branch 2 should going out over Branch 1. (Branch 2 is not connected directly with a PSTN.) Thank you in advace Housi Mueller Click here to donate to the Hurricane Katrina relief effort. ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation sponsored by Easynews .com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Click here to donate to the Hurricane Katrina relief effort. ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with include
Hi, I put on sip.conf the following line #include sip.d/*.conf inside I have files like that provider1.conf provider2.conf But asterisk does not want to load it This is the error Sep 1 13:18:35 VERBOSE[8756]: == Parsing '/etc/asterisk/sip.d/*.conf': Sep 1 13:18:35 VERBOSE[8756]: == Parsing '/etc/asterisk/sip.d/*.conf': Not found (No such file or directory) this is the ls result [EMAIL PROTECTED] asterisk]# ls /etc/asterisk/sip.d/ -la total 13 drwxrwxrwx 2 asterisk asterisk 4096 Sep 1 13:06 ./ drwxr-xr-x 9 asterisk asterisk 4096 Sep 1 13:17 ../ -rwxrwxrwx 1 asterisk asterisk 276 Sep 1 13:06 provider1.conf* -rwxrwxrwx 1 asterisk asterisk 274 Sep 1 13:06 provider2.conf* ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with include
Sorry, I'm using the 1.0.9On 9/1/05, Kevin P. Fleming [EMAIL PROTECTED] wrote: Il Neofita wrote: Hi, I put on sip.conf the following line #include sip.d/*.confYou neglected to include the most important piece of information: whatversion of Asterisk you are using. ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TXFAX() status
Hi, I'm using a script in order to send out my faxes with the application txfax, therefore, I do not know how to see if the faxes are sent. Any idea? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC and cdrs
Thank you was that the problem. On 8/25/05, Darren Wiebe [EMAIL PROTECTED] wrote: When did you install it? Try running the update database function from the configure menu. Darren Wiebe [EMAIL PROTECTED] Il Neofita wrote: My installation of ASTCC does not update the cdrs tables . It is a problem of ASTCC or it is a configuration problem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calling Card Application
I am looking for a calling card application which is able to advise me during a call when the credit is almost finish. For examples 1 minute before the end of the credit. Thank you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC and cdrs
My installation of ASTCC does not update the cdrs tables . It is a problem of ASTCC or it is a configuration problem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P compatible
Why my X100P detect the ring after 3 o 4 seconds? The funny thing that when I have an incoming call asterisk receive a signal but the commands start after 3 or 4 seconds. Moreover, when the call end the hungup has the same delay. any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TAFM
Hi, I installed this program but I am not able to configure, it does not want to work. Someone can help me? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TAFM
Hi, I checked but I did not find any info regarding the config files. I tough that I configured everything in the right way but I am not able to see anything on the web page. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
